similar to: multiple iax users on the same host

Displaying 20 results from an estimated 50000 matches similar to: "multiple iax users on the same host"

2007 Apr 21
3
FAX on PRI and TE205P
Hi i have a PRI connected to a TE205P. Actually, can i send and receive FAX through Asterisk using stable solutions? Or shall i connect an ATA to Asterisk and then a modem with Hylafax? -- /*************/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/nikstresser
2008 Jun 14
1
play sound on a specific channel
Hi to all can i play a sound or a dtmf tone on a specific channel using AMI? Thanks to all -- /*************/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser
2007 Sep 05
1
Spawn extension (default, 1002, 2) exited non-zero on 'SIP/host-0819d0d0
Hi i generate a call from the dialplan in this mode: exten => 1002,1,Answer() exten => 1002,2,Dial(SIP/user at host) the call is generated, but after some seconds it is interrupted, here the asterisk log: *CLI> -- Executing Answer("SIP/host1-0819d0d0", "") in new stack -- Executing Dial("SIP/host1-0819d0d0", "SIP/caller at host") in new
2007 May 07
2
h323 problem with asterisk 1.2.18
i am experiencing problem with asterisk 1.2.18 I've downloaded and installed pwlib and openh323 with the following commands: cd /path/to/pwlib ./configure make clean opt cd /path/to/openh323 ./configure make clean opt then 'ive set the corresponding PATH PWLIBDIR=/data/programmi/asterisk_1.2.18/pwlib_v1_10_0/ export PWLIBDIR OPENH323DIR=/data/programmi/asterisk_1.2.18/openh323_v1_18_0/
2007 Oct 27
0
Call center manager for Asterisk (Release 0.5)
CCMANAGER 0.5 released!! NOTE: this is a previous alpha release, maybe there is some customization to do on the settings files, i can't write a clear and complete howto at the moment I don't have released upgrades in the last months but the project is still alive i'm too busy at the moment, i'm following other projects to have some resources (both money and time) and then i can
2007 Aug 03
2
partial ChanSpy
Hi is it possible to spy (not record, spy) partially on a channel? for exaple, i'd like to listen only the input or the output voice. is it possible? thanks -- /*************/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/nikstresser
2008 Jul 17
1
OpenH323 and ptlib version for asterisk 1.4.21.1
Hi what version of openh323 and pwlib are suggested for asterisk 1.4.21.1.? Thanks to all -- /*************/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser
2007 Dec 23
3
OpenVox A800P01 and ZT_CHANCONFIG failed
Hi i've got an openvox a800p01 with 1 FXO and 4 FSX i've done the following: - downloaded zaptel-1.4.7.1 > >> - downloaded the file opvxa1200.c > >> - copied in zaptel-1.4.7.1/ > >> - edited makefile adding opvxa1200 in the modules and the voice > >> opvxa1200.o : zaptel.h wctdm.h > >> - edited zaptel.sysconfig adding MODULES="$MODULES
2007 Jun 29
4
asterisk call unique id in dialplan
Hi how can i retrieve the call unique id of asterisk in the dialplan? I have enabled the cdr logging on a postgres database. In the table cdr each record has a field that assumes an unique id (for example: 1141628669.51) Can i retrieve this from the dialplan? For example: exten => 203,1,Answer exten => 203,2,Set(CALLERID(name)=UNIQUE_ID - ${var_name_unique_id}) exten =>
2007 Jun 19
2
PhpAgi call generation
hi i'd like to write a simply application in php with phpAgi that: - connect to Asterisk - call an external number using a Zap channel - play a message here is some code: <?php $asm = new AGI_AsteriskManager(); if ($asm->connect()) { $asm->Originate("Zap/g1/1","number","default","1"); /* play message... */ } else {
2007 Nov 20
1
store 2 separate records in cdr when a call is transferd
Hi i've read this post http://lists.digium.com/pipermail/asterisk-dev/2007-May/027666.html I just want to know if there are some upgrades... on 1.4 or 1.2. I'd like to store two records in the CDR instead of one, when a call is transferd. Is it possibile now? Thanks to all -- /*************/ nik600 https://sourceforge.net/projects/ccmanager
2008 Jan 08
2
disable call waiting by default
I've connected some analogic phone to some fxs modules on an analogic card. I want to disable by default the call waiting sound. I know that dialing *70 before to call the call waiting is disabled until the next call, but isn't there a setting or a dialplan command to set up this automatically? Thanks -- /*************/ nik600 https://sourceforge.net/projects/ccmanager
2007 May 10
2
force outgoinc callerid
Hi i have a Te205P connected to a PRI E1, can i force the outgoing callerid to change for each context? for example: [outgoing_context_one] ;force callerid to 12345 exten => _XXXXXXXXXXX,1,Dial(Zap/${EXTEN}) [outgoing_context_two] ;force callerid to 22222 exten => _XXXXXXXXXXX,1,Dial(Zap/${EXTEN}) Can i do that? thanks to all -- /*************/ nik600
2007 Dec 11
0
new Asterisk installation with openvox 1.2 or 1.4?
Hi i need to install a server with this hardware: 1 OpenVox B800P 1 OpenVox A800P01 4 OpenVox FXS-100 FXS100 4 OctWare SoftEcho SOFTECHO Do you suggest 1.2 or 1.4 branch? Is now 1.4 stable ? I've tried 1.4 the last year but i've experienced many problems with misdn drivers. Thanks -- /*************/ nik600 https://sourceforge.net/projects/ccmanager
2008 Jul 09
0
disable DTMF on a particular channel
Hi to all is it possibile (via AMI or dialplan) to disable the DTMF tone on a particular channel? Thanks in advance -- /*************/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser
2008 Jun 04
0
Patch for app_asr.c: DTMF instead of goto
Hi to all if someone of you is interested on it, i've changed the code of app_asr.c With these patch you can use the ASR application to play DTMF tones, so you can have your own AGI application that uses the ASR and manages the DTMF tones without change the dialplan. EXAMPLE exten => 003,1,Ringing exten => 003,2,Wait(3) exten => 003,3,Answer exten =>
2007 Sep 13
0
[phpAGI] generate a call from a SIP device to Asterisk
Hi i need to generate a call from a SIP hardware device to Asterisk. The device isn't registered with a sip account to Asterisk. What i've done, is to do this (using phpAGI): ..... $asm->Originate(SIP/user_on_device at ip_of_device,2000,"default","1"); ..... And on the extension 2000 in the context "default" exten => 2000,1,ChanSpy(|g(100)) exten
2006 Apr 19
2
Asterisk 1.2.7.1 and IAX modem / channel
Hi, I was using Asterisk with Hylafax via IAX Modem. It works fine until I upgraded to Asterisk 1.2.7.1 I didn't change any configuration but it seems that Asterisk does not get the call from IAXModem anymore. I'm doing something like this Asterisk <--> IAXModem <--> Hylafax Usually when I use sendfax -n -d 260XXX somefile I'll see Asterisk receiving the call in
2008 Jul 01
4
Fax Between IAX Trunks
Hello! I need to send Faxes from an Asterisk box to an Asterisk + Iaxmodem + Hylafax installed on other box. I have setup IAX trunks between this boxes, all works fine but can?t send faxes from one to other, Im trying with or without NVFaxDetect application but does not work. Is there a way to get it working?. If I connect a fax machine directly to Asterisk with Iaxmodem and Hylafax, I have no
2007 Feb 16
1
iaxmodem - fax tone?
I am testing out hylafax and iaxmodem. Submitting jobs to the hylafax server and having it then dial using iaxmodem is working fine. Hylafax server is talking to my Asterisk box that has a Sangoma A101 in it via iaxmodem via an IAX channel using ulaw. A call coming into a certain test DID comes into the Sangoma A101 then it goes to another box via IAX ulaw that uses the rxfax app to